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3e9e5b39d1
So that we don't have to be capable of creating one ourselves, which requires a dependency on the audio codecs. BUG=webrtc:8396 Change-Id: I5600da5e17f613b0e61a9fb0fbdb373fe42f855c Reviewed-on: https://webrtc-review.googlesource.com/20220 Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20641}
198 lines
9 KiB
C++
198 lines
9 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "media/base/fakemediaengine.h"
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#include "ortc/ortcfactory.h"
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#include "ortc/testrtpparameters.h"
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#include "p2p/base/fakepackettransport.h"
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#include "rtc_base/gunit.h"
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namespace webrtc {
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// This test uses fake packet transports and a fake media engine, in order to
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// test the RtpTransportController at only an API level. Any end-to-end test
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// should go in ortcfactory_integrationtest.cc instead.
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//
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// Currently, this test mainly focuses on the limitations of the "adapter"
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// RtpTransportController implementation. Only one of each type of
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// sender/receiver can be created, and the sender/receiver of the same media
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// type must use the same transport.
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class RtpTransportControllerTest : public testing::Test {
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public:
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RtpTransportControllerTest() {
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// Note: This doesn't need to use fake network classes, since it uses
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// FakePacketTransports.
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auto result = OrtcFactory::Create(
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nullptr, nullptr, nullptr, nullptr, nullptr,
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std::unique_ptr<cricket::MediaEngineInterface>(
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new cricket::FakeMediaEngine()),
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CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
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ortc_factory_ = result.MoveValue();
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rtp_transport_controller_ =
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ortc_factory_->CreateRtpTransportController().MoveValue();
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}
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protected:
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std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
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std::unique_ptr<RtpTransportControllerInterface> rtp_transport_controller_;
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};
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TEST_F(RtpTransportControllerTest, GetTransports) {
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rtc::FakePacketTransport packet_transport1("one");
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rtc::FakePacketTransport packet_transport2("two");
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auto rtp_transport_result1 = ortc_factory_->CreateRtpTransport(
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MakeRtcpMuxParameters(), &packet_transport1, nullptr,
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rtp_transport_controller_.get());
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ASSERT_TRUE(rtp_transport_result1.ok());
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auto rtp_transport_result2 = ortc_factory_->CreateRtpTransport(
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MakeRtcpMuxParameters(), &packet_transport2, nullptr,
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rtp_transport_controller_.get());
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ASSERT_TRUE(rtp_transport_result2.ok());
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auto returned_transports = rtp_transport_controller_->GetTransports();
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ASSERT_EQ(2u, returned_transports.size());
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EXPECT_EQ(rtp_transport_result1.value().get(), returned_transports[0]);
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EXPECT_EQ(rtp_transport_result2.value().get(), returned_transports[1]);
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// If a transport is deleted, it shouldn't be returned any more.
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rtp_transport_result1.MoveValue().reset();
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returned_transports = rtp_transport_controller_->GetTransports();
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ASSERT_EQ(1u, returned_transports.size());
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EXPECT_EQ(rtp_transport_result2.value().get(), returned_transports[0]);
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}
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// Create RtpSenders and RtpReceivers on top of RtpTransports controlled by the
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// same RtpTransportController. Currently only one each of audio/video is
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// supported.
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TEST_F(RtpTransportControllerTest, AttachMultipleSendersAndReceivers) {
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rtc::FakePacketTransport audio_packet_transport("audio");
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rtc::FakePacketTransport video_packet_transport("video");
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auto audio_rtp_transport_result = ortc_factory_->CreateRtpTransport(
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MakeRtcpMuxParameters(), &audio_packet_transport, nullptr,
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rtp_transport_controller_.get());
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ASSERT_TRUE(audio_rtp_transport_result.ok());
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auto audio_rtp_transport = audio_rtp_transport_result.MoveValue();
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auto video_rtp_transport_result = ortc_factory_->CreateRtpTransport(
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MakeRtcpMuxParameters(), &video_packet_transport, nullptr,
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rtp_transport_controller_.get());
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ASSERT_TRUE(video_rtp_transport_result.ok());
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auto video_rtp_transport = video_rtp_transport_result.MoveValue();
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auto audio_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
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EXPECT_TRUE(audio_sender_result.ok());
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auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
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EXPECT_TRUE(audio_receiver_result.ok());
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auto video_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
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EXPECT_TRUE(video_sender_result.ok());
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auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
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EXPECT_TRUE(video_receiver_result.ok());
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// Now that we have one each of audio/video senders/receivers, trying to
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// create more on top of the same controller is expected to fail.
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// TODO(deadbeef): Update this test once multiple senders/receivers on top of
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// the same controller is supported.
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auto failed_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
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EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
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failed_sender_result.error().type());
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auto failed_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
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EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
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failed_receiver_result.error().type());
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failed_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
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EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
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failed_sender_result.error().type());
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failed_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
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EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
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failed_receiver_result.error().type());
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// If we destroy the existing sender/receiver using a transport controller,
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// we should be able to make a new one, despite the above limitation.
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audio_sender_result.MoveValue().reset();
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audio_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
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EXPECT_TRUE(audio_sender_result.ok());
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audio_receiver_result.MoveValue().reset();
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audio_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
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EXPECT_TRUE(audio_receiver_result.ok());
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video_sender_result.MoveValue().reset();
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video_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
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EXPECT_TRUE(video_sender_result.ok());
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video_receiver_result.MoveValue().reset();
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video_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
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EXPECT_TRUE(video_receiver_result.ok());
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}
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// Given the current limitations of the BaseChannel-based implementation, it's
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// not possible for an audio sender and receiver to use different RtpTransports.
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// TODO(deadbeef): Once this is supported, update/replace this test.
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TEST_F(RtpTransportControllerTest,
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SenderAndReceiverUsingDifferentTransportsUnsupported) {
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rtc::FakePacketTransport packet_transport1("one");
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rtc::FakePacketTransport packet_transport2("two");
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auto rtp_transport_result1 = ortc_factory_->CreateRtpTransport(
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MakeRtcpMuxParameters(), &packet_transport1, nullptr,
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rtp_transport_controller_.get());
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ASSERT_TRUE(rtp_transport_result1.ok());
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auto rtp_transport1 = rtp_transport_result1.MoveValue();
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auto rtp_transport_result2 = ortc_factory_->CreateRtpTransport(
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MakeRtcpMuxParameters(), &packet_transport2, nullptr,
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rtp_transport_controller_.get());
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ASSERT_TRUE(rtp_transport_result2.ok());
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auto rtp_transport2 = rtp_transport_result2.MoveValue();
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// Create an audio sender on transport 1, then try to create a receiver on 2.
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auto audio_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_AUDIO, rtp_transport1.get());
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EXPECT_TRUE(audio_sender_result.ok());
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auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_AUDIO, rtp_transport2.get());
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EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
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audio_receiver_result.error().type());
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// Delete the sender; now we should be ok to create the receiver on 2.
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audio_sender_result.MoveValue().reset();
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audio_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_AUDIO, rtp_transport2.get());
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EXPECT_TRUE(audio_receiver_result.ok());
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// Do the same thing for video, reversing 1 and 2 (for variety).
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auto video_sender_result = ortc_factory_->CreateRtpSender(
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cricket::MEDIA_TYPE_VIDEO, rtp_transport2.get());
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EXPECT_TRUE(video_sender_result.ok());
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auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
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EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
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video_receiver_result.error().type());
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video_sender_result.MoveValue().reset();
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video_receiver_result = ortc_factory_->CreateRtpReceiver(
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cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
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EXPECT_TRUE(video_receiver_result.ok());
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}
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} // namespace webrtc
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